AudioCodes Mediant 4000 SBC User Manual page 444

Session border controllers
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CHAPTER 19    Coders and Profiles
Parameter
'Jitter Compensation'
sbc-jitter-compensation
[IpProfile_
SBCJitterCompensation]
'ICE Mode'
ice-mode
[IPProfile_SBCIceMode]
'SDP Handle RTCP'
sbc-sdp-handle-rtcp
[IpProfile_
SBCSDPHandleRTCPAttribut
e]
Description
Enables the on-demand jitter buffer for SBC calls. The jitter
buffer can be used when other functionality such as voice
transcoding are not done on the call. The jitter buffer is
useful when incoming packets are received at inconsistent
intervals (i.e., packet delay variation). The jitter buffer
stores the packets and sends them out at a constant rate
(according to the coder's settings).
[0] Disable (default)
[1] Enable
Note:
The jitter buffer parameters, 'Dynamic Jitter Buffer
Minimum Delay' (DJBufMinDelay) and 'Dynamic Jitter
Buffer Optimization Factor' (DJBufOptFactor) can be
used to configure minimum packet delay only when
transcoding is employed.
This functionality may require DSP resources. For
more information, contact the sales representative of
your purchased device.
Enables Interactive Connectivity Establishment (ICE) Lite
for the SIP entity associated with the IP Profile. ICE is a
methodology for NAT traversal, employing the Session
Traversal Utilities for NAT (STUN) and Traversal Using
Relays around NAT (TURN) protocols to provide a peer
with a public IP address and port that can be used to
connect to a remote peer.
[0] Disable (default)
[1] Lite
For more information on ICE Lite, see
Note: As ICE has been defined by the WebRTC standard
as mandatory, the support is important for deployments
implementing WebRTC. For more information on
WebRTC, see WebRTC.
Enables the interworking of the RTCP attribute, 'a=rtcp'
(RTCP) in the SDP, for the SIP entity associated with the
IP Profile. The RTCP attribute is used to indicate the
RTCP port for media when that port is not the next higher
port number following the RTP port specified in the media
line ('m=').
The parameter is useful for SIP entities that either require
the attribute or do not support the attribute. For example,
Google Chrome and Web RTC do not accept calls without
the RTCP attribute in the SDP. In Web RTC, Chrome
(SDES) generates the SDP with 'a=rtcp', for example:
m=audio 49170 RTP/AVP 0
a=rtcp:53020 IN IP6
2001:2345:6789:ABCD:EF01:2345:6789:ABCD
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Mediant 4000 SBC | User's Manual
ICE
Lite.

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